bitflipper

The reason I asked is because my band's singer had asked me if it was possible. She's jonesin' for some quality band time but of course like everybody else around these parts we are all self-quarantined. No gigs, no rehearsals. Personally, I'm content jamming alone, but that's not an option for the singer nor her drummer husband. I'd told her I didn't think it was possible over the internet, but researched it anyway. I ran across this thread on Gearslutz where it was being discussed. I checked out the free open source software but didn't download it because I'm still convinced it isn't practical. Maybe on a local area network, but not over the internet. I stopped short of recommending a conventional online collaboration, given that of this six-piece band only one other member is conversant in DAW-speak.

How are they doing this? I had assumed that online jam software over large physical distances was inherently impractical due to the unavoidable network latency.

It'll work. However, the reason I use a smaller monitor (34") is that anything wider is going to impinge on line of sight to the monitors and skew their frequency response. If that's not enough screen real estate, get two and mount them one above the other. My older 24" screens are mounted on an arm and sit above the 34" display. One is used for plugins and browser, the other for the PRV or Melodyne.

Create a separate cue mix bus and put the reverb there. The singer will hear it in the headphones but it won't be recorded. Caveat: any software reverb is going to introduce latency that can make the vocal timing sound sloppy. I'd recommend buying a cheap hardware reverb unit just for vocal monitoring.

So the reason you can't use the configuration that works (ASIO) is that there are problems with 96 KHz and above? Is there a reason why you need to run the interface at 96 KHz? Does Pianoteq sound better at the higher rate, or are you just doing it for the lower latency?

Try to give legit acoustical advice and this is the treatment I get.

I work at home anyway, but I have little to do work-wise since most of my customers are shut down. And because I can't bring myself to bill anyone under these circumstances, I probably won't see any income at all for months to come. So no musical purchases this year. Unfortunately, I'd recently spent a couple grand on keyboard amplification that I now won't get to use, given that there aren't any gigs. I should have spent the money on toilet paper instead. I imagine 5,000 rolls stacked in the corners would make great bass traps.

Clean Audio Folder removes files that are no longer referenced. User 905133's problem is the opposite: there is a reference but no matching file.

In that case, there is still a (bogus) link to some nonexistent clip in one of your tracks. Unfortunately, there is no convenient way to identify which track it is. Here's how I handled that scenario when it happened to me... First, make a copy of the entire project folder. This will be a temporary workspace for diagnostics and you won't need it afterward. Now open that copy and start deleting tracks and saving the project. Repeat until it comes up without the missing audio warnings, and then you'll know which track was having the problem. In my case, I had some 60 tracks in the project so I used a binary approach to save time, deleting half the tracks, then half the remaining tracks and so on. Once you've identified the problem track, clone it but don't clone events (just plugins, automation and routing). Select all the clips in the original track and CTL-SHIFT-drag them down into the new track, which will be clean of orphaned links. Finally, delete the original problem track and away you go.

Your audio interface plays no role in exports, unless you're using external inserts. First thing I'd do is import one of those files back into the project and verify that its level has really changed, as opposed to being altered by SoundForge. Just to be sure, you're exporting these tracks as 24- or 32-bit wave files, right?

If you're looking specifically for a versatile envelope-follower/modulatable filter, I know of nothing else that's in Volcano's league. Certainly not any freebies. Main downside to Volcano is it's rather complicated if you want to dive deep. Meldaproduction has a simple but pretty powerful one (MFilter), but I'd wait for it to go on sale for $27. You can demo it for free.

You can't draw conclusions from statistics you don't have. In the US, we have tested less than 0.2% of the population for the virus. Based on sample sizes like that, one could conclude that not only are sharks not dangerous, they probably don't even exist. Ask anyone in Nebraska if they've ever seen one.

Are these mono tracks being exported as mono waves?

I sure hope 5-pin DINs and UARTs don't ever go away. They are superior to USB in almost every regard. As to whether hardware always beats software, that's another can o' worms. Digital hardware synths will usually have design restrictions that a computer needn't be subject to, such as reduced storage space and slower CPUs. The supposed superiority of analog synths is a moot point - those pristine waveforms are going to end up as digital data eventually. I have found balance by using both hardware and software. I play a high-end digital synthesizer on stage, but in the studio it's solely a MIDI controller. I can get any kind of sound I want from it while playing/recording, thus avoiding the need to ever play a software synth in real time. Consequently, latency is never a concern despite keeping my buffers at their highest setting all the time.

Almost no USB MIDI here. I have a MIDI multiplexer, a sort of 5-pin DIN patch bay with presets. Also some standalone splitters and a combiner so I can drive a common MIDI device from two keyboards onstage. My only USB connection is to a tiny portable keyboard that fits into my laptop bag. Its only purpose is to give me something to do in airport waiting lounges. No need to chain it to anything.

Maybe this is the true value of 432 Hz tuning. I've settled on 433.5 Hz just to be safe. Empirical testing has shown that to let in just enough cosmic vibrations and still fall outside the parameters of automated melody detection algorithms.

^^^ This is my logic, exactly. I'm self-quarantining. But I've been soaking my microphones in alcohol just to be safe. The surgical mask over the pop filter didn't work out so well.

Step one: verify that the DLL is actually where you think it is. Yeh, that falls under the "duh" category but it's always gotta be step 1. I've been bitten by not making it step #1, e.g. the plugin's installer put it somewhere I didn't expect. Once you've verified the DLL is present and where you think it should be, there are only two reasons for it to not show up: 1. The VST's internal registration failed, or 2. Cakewalk never initiated the registration because it didn't know about the VST's existence. Open a scan log and re-scan all your VSTs. Do a text search within the resulting log file on the DLL's filename. If it's not there at all, then it's scenario #2 (you haven't told the scanner where to find the DLL). If it's in the log but raised an error during initialization, the error may provide a clue as to what went wrong, e.g. a missing dependency or insufficient Windows permissions to create some file.

If this was a paid gig, I'd say go out and buy iZotope Rx. But given that it's a freebie, Adobe Audition should do OK. It has decent noise reduction features, and its spectral editor can handle the one-off noises (be ready to spend some hours with your headphones on). As Jim points out, it will probably require multiple passes. Fix the noise first, then use LP and HP filters to narrow the spectrum to just the speech frequencies, and finally work on the amplitude issues with gain automation, gating and compression.

That's mod wheel data. Cakewalk isn't sending it, it's your controller. Or (less likely) your MIDI interface.

You could say the same about 12-string guitars.

The band above is Nightwish, a mostly-Finnish band, although the pipe player Troy Donocklay is British (their current singer, not shown, is Dutch). One of my favorite bands in the world, certainly my absolute favorite Scandinavian metal band. And yes, his instruments are real. Interesting you should mention Gary Moore, as Nightwish covered a Moore tune "Over the Hills and Far Away" and it's one of their most-requested live numbers.

^^^ Craig has succinctly identified the main advantage of the K-system over LUFS: the former is built around monitor calibration while the latter ignores the problem of subjectivity in the mix in an attempt to make the process a completely objective exercise. They are really two different things, even though they have similar goals. LUFS solves the longstanding need for standardization in broadcasting and streaming, but it does not help you get a good mix to begin with. For that you need to monitor at consistent levels, and that's a prerequisite for implementing the K-system. I use both systems. K-14 or K-20 while mixing, LUFS for the final master. If I have to alter overall volume to hit the LUFS target, monitoring at the same levels as dictated by the K-system helps me hear whether that volume change has made it necessary to tweak the mix.

As long as you don't modify the beginning of a wave file, an external edit should not affect the file's position on the timeline, as that information is not contained in the wave file itself but rather in the project. I've been using an ancient copy of Adobe Audition this way for many years, and have never had an alignment issue. However, I do 99% of my editing within Cakewalk itself, using splits, slip edits, mutes and automation to fix most problems. An external editor only needs to come into play if I'm trying to remove noise, as CW does not have a de-noiser or de-clicker utility. Those are specialty tools best left to dedicated software such as iZotope Rx and Sony SoundForge.